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Bug ID	13193
Summary	Support for DTLS-SRTP (used by WebRTC) with SIP signalling over Websockets
Product	Wireshark
Version	Git
Hardware	All
OS	All
Status	UNCONFIRMED
Severity	Enhancement
Priority	Low
Component	Dissection engine (libwireshark)
Assignee	bugzilla-admin@wireshark.org
Reporter	peter@lekensteyn.nl

Created attachment 15095 [details]
Sample WSS/SIP + DTLS-SRTP/G.711 capture (Firefox 50, Asterisk 11.13.1)

Build Information:

--
WebRTC is a platform for web developer that want to implement realtime
multimedia applications. Its architecture is described at
https://webrtc.org/architecture/

This bug tracks the required changes to improve WebRTC support:
- SIP over Websockets (RFC 7118) (fixed today via bug 11420)
- STUN/ICE: need to extend SDP dissector to handle candidates
- DTLS-SRTP (RFC 5764): add use_srtp DTLS extension and add key extractor
- RTP: decryption support for SRTP is missing
- Support for playing the Opus audio codec (MTI per RFC 7874)
- ...

Refs:

Interactive Connectivity Establishment (ICE): A Protocol for Network Address
Translator (NAT) Traversal for Offer/Answer Protocols
https://tools.ietf.org/html/rfc5245

The Secure Real-time Transport Protocol (SRTP)
https://tools.ietf.org/html/rfc3711

Framework for Establishing a Secure Real-time Transport Protocol (SRTP)
Security Context Using Datagram Transport Layer Security (DTLS)
https://tools.ietf.org/html/rfc5763

Datagram Transport Layer Security (DTLS) Extension to Establish Keys
for the Secure Real-time Transport Protocol (SRTP)
https://tools.ietf.org/html/rfc5764

WebRTC Audio Codec and Processing Requirements
https://tools.ietf.org/html/rfc7874

RTP Payload Format for the Opus Speech and Audio Codec
https://tools.ietf.org/html/rfc7587

Multiplexing RTP Data and Control Packets on a Single Port
https://tools.ietf.org/html/rfc5761

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